Network Working Group T. Friedman, Ed.
Request for Comments: 3611 Paris 6
Category: Standards Track R. Caceres, Ed.
IBM Research
A. Clark, Ed.
Telchemy
November 2003
RTP Control Protocol Extended Reports (RTCP XR)
Status of this Memo
This document specifies an Internet standards track protocol for the
Internet community, and requests discussion and suggestions for
improvements. Please refer to the current edition of the "Internet
Official Protocol Standards" (STD 1) for the standardization state
and status of this protocol. Distribution of this memo is unlimited.
Copyright Notice
Copyright (C) The Internet Society (2003). All Rights Reserved.
Abstract
This document defines the Extended Report (XR) packet type for the
RTP Control Protocol (RTCP), and defines how the use of XR packets
can be signaled by an application if it employs the Session
Description Protocol (SDP). XR packets are composed of report
blocks, and seven block types are defined here. The purpose of the
extended reporting format is to convey information that supplements
the six statistics that are contained in the report blocks used by
RTCP's Sender Report (SR) and Receiver Report (RR) packets. Some
applications, such as multicast inference of network characteristics
(MINC) or voice over IP (VoIP) monitoring, require other and more
detailed statistics. In addition to the block types defined here,
additional block types may be defined in the future by adhering to
the framework that this document provides.
Friedman, et al. Standards Track [Page 1]

RFC 3611 RTCP XR November 20031. Introduction
This document defines the Extended Report (XR) packet type for the
RTP Control Protocol (RTCP) [9], and defines how the use of XR
packets can be signaled by an application if it employs the Session
Description Protocol (SDP) [4]. XR packets convey information beyond
that already contained in the reception report blocks of RTCP's
sender report (SR) or Receiver Report (RR) packets. The information
is of use across RTP profiles, and so is not appropriately carried in
SR or RR profile-specific extensions. Information used for network
management falls into this category, for instance.
The definition is broken out over the three sections that follow the
Introduction. Section 2 defines the XR packet as consisting of an
eight octet header followed by a series of components called report
blocks. Section 3 defines the common format, or framework,
consisting of a type and a length field, required for all report
blocks. Section 4 defines several specific report block types.
Other block types can be defined in future documents as the need
arises.
The report block types defined in this document fall into three
categories. The first category consists of packet-by-packet reports
on received or lost RTP packets. Reports in the second category
convey reference time information between RTP participants. In the
third category, reports convey metrics relating to packet receipts,
that are summary in nature but that are more detailed, or of a
different type, than that conveyed in existing RTCP packets.
All told, seven report block formats are defined by this document.
Of these, three are packet-by-packet block types:
- Loss RLE Report Block (Section 4.1): Run length encoding of
reports concerning the losses and receipts of RTP packets.
- Duplicate RLE Report Block (Section 4.2): Run length encoding of
reports concerning duplicates of received RTP packets.
- Packet Receipt Times Report Block (Section 4.3): A list of
reception timestamps of RTP packets.
There are two reference time related block types:
- Receiver Reference Time Report Block (Section 4.4): Receiver-end
wallclock timestamps. Together with the DLRR Report Block
mentioned next, these allow non-senders to calculate round-trip
times.
Friedman, et al. Standards Track [Page 3]

RFC 3611 RTCP XR November 2003
- DLRR Report Block (Section 4.5): The delay since the last Receiver
Reference Time Report Block was received. An RTP data sender that
receives a Receiver Reference Time Report Block can respond with a
DLRR Report Block, in much the same way as, in the mechanism
already defined for RTCP [9, Section 6.3.1], an RTP data receiver
that receives a sender's NTP timestamp can respond by filling in
the DLSR field of an RTCP reception report block.
Finally, this document defines two summary metric block types:
- Statistics Summary Report Block (Section 4.6): Statistics on RTP
packet sequence numbers, losses, duplicates, jitter, and TTL or
Hop Limit values.
- VoIP Metrics Report Block (Section 4.7): Metrics for monitoring
Voice over IP (VoIP) calls.
Before proceeding to the XR packet and report block definitions, this
document provides an applicability statement (Section 1.1) that
describes the contexts in which these report blocks can be used. It
also defines (Section 1.2) the normative use of key words, such as
MUST and SHOULD, as they are employed in this document.
Following the definitions of the various report blocks, this document
describes how applications that employ SDP can signal their use
(Section 5). The document concludes with a discussion (Section 6) of
numbering considerations for the Internet Assigned Numbers Authority
(IANA), of security considerations (Section 7), and with appendices
that provide examples of how to implement algorithms discussed in the
text.
1.1. Applicability
The XR packets are useful across multiple applications, and for that
reason are not defined as profile-specific extensions to RTCP sender
or Receiver Reports [9, Section 6.4.3]. Nonetheless, they are not of
use in all contexts. In particular, the VoIP metrics report block
(Section 4.7) is specific to voice applications, though it can be
employed over a wide variety of such applications.
The VoIP metrics report block can be applied to any one-to-one or
one-to-many voice application for which the use of RTP and RTCP is
specified. The use of conversational metrics (Section 4.7.5),
including the R factor (as described by the E Model defined in [3])
and the mean opinion score for conversational quality (MOS-CQ), in
applications other than simple two party calls is not defined; hence,
these metrics should be identified as unavailable in multicast
conferencing applications.
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RFC 3611 RTCP XR November 2003
The packet-by-packet report block types, Loss RLE (Section 4.1),
Duplicate RLE (Section 4.2), and Packet Receipt Times (Section 4.3),
have been defined with network tomography applications, such as
multicast inference of network characteristics (MINC) [11], in mind.
MINC requires detailed packet receipt traces from multicast session
receivers in order to infer the gross structure of the multicast
distribution tree and the parameters, such as loss rates and delays,
that apply to paths between the branching points of that tree.
Any real time multicast multimedia application can use the packet-
by-packet report block types. Such an application could employ a
MINC inference subsystem that would provide it with multicast tree
topology information. One potential use of such a subsystem would be
for the identification of high loss regions in the multicast tree and
the identification of multicast session participants well situated to
provide retransmissions of lost packets.
Detailed packet-by-packet reports do not necessarily have to consume
disproportionate bandwidth with respect to other RTCP packets. An
application can cap the size of these blocks. A mechanism called
"thinning" is provided for these report blocks, and can be used to
ensure that they adhere to a size limit by restricting the number of
packets reported upon within any sequence number interval. The
rationale for, and use of this mechanism is described in [13].
Furthermore, applications might not require report blocks from all
receivers in order to answer such important questions as where in the
multicast tree there are paths that exceed a defined loss rate
threshold. Intelligent decisions regarding which receivers send
these report blocks can further restrict the portion of RTCP
bandwidth that they consume.
The packet-by-packet report blocks can also be used by dedicated
network monitoring applications. For such an application, it might
be appropriate to allow more than 5% of RTP data bandwidth to be used
for RTCP packets, thus allowing proportionately larger and more
detailed report blocks.
Nothing in the packet-by-packet block types restricts their use to
multicast applications. In particular, they could be used for
network tomography similar to MINC, but using striped unicast packets
instead. In addition, if it were found useful, they could be used
for applications limited to two participants.
One use to which the packet-by-packet reports are not immediately
suited is for data packet acknowledgments as part of a packet
retransmission mechanism. The reason is that the packet accounting
technique suggested for these blocks differs from the packet
accounting normally employed by RTP. In order to favor measurement
Friedman, et al. Standards Track [Page 5]

RFC 3611 RTCP XR November 2003
applications, an effort is made to interpret as little as possible at
the data receiver, and leave the interpretation as much as possible
to participants that receive the report blocks. Thus, for example, a
packet with an anomalous SSRC ID or an anomalous sequence number
might be excluded by normal RTP accounting, but would be reported
upon for network monitoring purposes.
The Statistics Summary Report Block (Section 4.6) has also been
defined with network monitoring in mind. This block type can be used
equally well for reporting on unicast and multicast packet reception.
The reference time related block types were conceived for receiver-
based TCP-friendly multicast congestion control [18]. By allowing
data receivers to calculate their round trip times to senders, they
help the receivers estimate the downstream bandwidth they should
request. Note that if every receiver is to send Receiver Reference
Time Report Blocks (Section 4.4), a sender might potentially send a
number of DLRR Report Blocks (Section 4.5) equal to the number of
receivers whose RTCP packets have arrived at the sender within its
reporting interval. As the number of participants in a multicast
session increases, an application should use discretion regarding
which participants send these blocks, and how frequently.
XR packets supplement the existing RTCP packets, and may be stacked
with other RTCP packets to form compound RTCP packets [9, Section 6].
The introduction of XR packets into a session in no way changes the
rules governing the calculation of the RTCP reporting interval [9,
Section 6.2]. As XR packets are RTCP packets, they count as such for
bandwidth calculations. As a result, the addition of extended
reporting information may tend to increase the average RTCP packet
size, and thus the average reporting interval. This increase may be
limited by limiting the size of XR packets.
The SDP signaling defined for XR packets in this document (Section 5)
was done so with three use scenarios in mind: a Real Time Streaming
Protocol (RTSP) controlled streaming application, a one-to-many
multicast multimedia application such as a course lecture with
enhanced feedback, and a Session Initiation Protocol (SIP) controlled
conversational session involving two parties. Applications that
employ SDP are free to use additional SDP signaling for cases not
covered here. In addition, applications are free to use signaling
mechanisms other than SDP.
Friedman, et al. Standards Track [Page 6]

RFC 3611 RTCP XR November 20031.2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in BCP 14, RFC 2119 [1]
and indicate requirement levels for compliance with this
specification.
2. XR Packet Format
An XR packet consists of a header of two 32-bit words, followed by a
number, possibly zero, of extended report blocks. This type of
packet is laid out in a manner consistent with other RTCP packets, as
concerns the essential version, packet type, and length information.
XR packets are thus backwards compatible with RTCP receiver
implementations that do not recognize them, but that ought to be able
to parse past them using the length information. A padding field and
an SSRC field are also provided in the same locations that they
appear in other RTCP packets, for simplicity. The format is as
follows:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P|reserved | PT=XR=207 | length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: report blocks :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
version (V): 2 bits
Identifies the version of RTP. This specification applies to
RTP version two.
padding (P): 1 bit
If the padding bit is set, this XR packet contains some
additional padding octets at the end. The semantics of this
field are identical to the semantics of the padding field in
the SR packet, as defined by the RTP specification.
reserved: 5 bits
This field is reserved for future definition. In the absence
of such definition, the bits in this field MUST be set to zero
and MUST be ignored by the receiver.
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RFC 3611 RTCP XR November 2003
packet type (PT): 8 bits
Contains the constant 207 to identify this as an RTCP XR
packet. This value is registered with the Internet Assigned
Numbers Authority (IANA), as described in Section 6.1.
length: 16 bits
As described for the RTCP Sender Report (SR) packet (see
Section 6.4.1 of the RTP specification [9]). Briefly, the
length of this XR packet in 32-bit words minus one, including
the header and any padding.
SSRC: 32 bits
The synchronization source identifier for the originator of
this XR packet.
report blocks: variable length.
Zero or more extended report blocks. In keeping with the
extended report block framework defined below, each block MUST
consist of one or more 32-bit words.
3. Extended Report Block Framework
Extended report blocks are stacked, one after the other, at the end
of an XR packet. An individual block's length is a multiple of 4
octets. The XR header's length field describes the total length of
the packet, including these extended report blocks.
Each block has block type and length fields that facilitate parsing.
A receiving application can demultiplex the blocks based upon their
type, and can use the length information to locate each successive
block, even in the presence of block types it does not recognize.
An extended report block has the following format:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| BT | type-specific | block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: type-specific block contents :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
block type (BT): 8 bits
Identifies the block format. Seven block types are defined in
Section 4. Additional block types may be defined in future
specifications. This field's name space is managed by the
Internet Assigned Numbers Authority (IANA), as described in
Section 6.2.
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RFC 3611 RTCP XR November 2003
type-specific: 8 bits
The use of these bits is determined by the block type
definition.
block length: 16 bits
The length of this report block, including the header, in 32-
bit words minus one. If the block type definition permits,
zero is an acceptable value, signifying a block that consists
of only the BT, type-specific, and block length fields, with a
null type-specific block contents field.
type-specific block contents: variable length
The use of this field is defined by the particular block type,
subject to the constraint that it MUST be a multiple of 32 bits
long. If the block type definition permits, It MAY be zero
bits long.
4. Extended Report Blocks
This section defines seven extended report blocks: block types for
reporting upon received packet losses and duplicates, packet
reception times, receiver reference time information, receiver
inter-report delays, detailed reception statistics, and voice over IP
(VoIP) metrics. An implementation SHOULD ignore incoming blocks with
types not relevant or unknown to it. Additional block types MUST be
registered with the Internet Assigned Numbers Authority (IANA) [16],
as described in Section 6.2.
4.1. Loss RLE Report Block
This block type permits detailed reporting upon individual packet
receipt and loss events. Such reports can be used, for example, for
multicast inference of network characteristics (MINC) [11]. With
MINC, one can discover the topology of the multicast tree used for
distributing a source's RTP packets, and of the loss rates along
links within that tree, or they could be used to provide raw data to
a network management application.
Since a Boolean trace of lost and received RTP packets is potentially
lengthy, this block type permits the trace to be compressed through
run length encoding. To further reduce block size, loss event
reports can be systematically dropped from the trace in a mechanism
called thinning that is described below and that is studied in [13].
A participant that generates a Loss RLE Report Block should favor
accuracy in reporting on observed events over interpretation of those
events whenever possible. Interpretation should be left to those who
observe the report blocks. Following this approach implies that
Friedman, et al. Standards Track [Page 9]

RFC 3611 RTCP XR November 2003
accounting for Loss RLE Report Blocks will differ from the accounting
for the generation of the SR and RR packets described in the RTP
specification [9] in the following two areas: per-sender accounting
and per-packet accounting.
In its per-sender accounting, an RTP session participant SHOULD NOT
make the receipt of a threshold minimum number of RTP packets a
condition for reporting upon the sender of those packets. This
accounting technique differs from the technique described in Section6.2.1 and Appendix A.1 of the RTP specification that allows a
threshold to determine whether a sender is considered valid.
In its per-packet accounting, an RTP session participant SHOULD treat
all sequence numbers as valid. This accounting technique differs
from the technique described in Appendix A.1 of the RTP specification
that suggests ruling a sequence number valid or invalid on the basis
of its contiguity with the sequence numbers of previously received
packets.
Sender validity and sequence number validity are interpretations of
the raw data. Such interpretations are justified in the interest,
for example, of excluding the stray old packet from an unrelated
session from having an effect upon the calculation of the RTCP
transmission interval. The presence of stray packets might, on the
other hand, be of interest to a network monitoring application.
One accounting interpretation that is still necessary is for a
participant to decide whether the 16 bit sequence number has rolled
over. Under ordinary circumstances this is not a difficult task.
For example, if packet number 65,535 (the highest possible sequence
number) is followed shortly by packet number 0, it is reasonable to
assume that there has been a rollover. However, it is possible that
the packet is an earlier one (from 65,535 packets earlier). It is
also possible that the sequence numbers have rolled over multiple
times, either forward or backward. The interpretation becomes more
difficult when there are large gaps between the sequence numbers,
even accounting for rollover, and when there are long intervals
between received packets.
The per-packet accounting technique mandated here is for a
participant to keep track of the sequence number of the packet most
recently received from a sender. For the next packet that arrives
from that sender, the sequence number MUST be judged to fall no more
than 32,768 packets ahead or behind the most recent one, whichever
choice places it closer. In the event that both choices are equally
distant (only possible when the distance is 32,768), the choice MUST
be the one that does not require a rollover. Appendix A.1 presents
an algorithm that implements this technique.
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RFC 3611 RTCP XR November 2003
Each block reports on a single RTP data packet source, identified by
its SSRC. The receiver that is supplying the report is identified in
the header of the RTCP packet.
Choice of beginning and ending RTP packet sequence numbers for the
trace is left to the application. These values are reported in the
block. The last sequence number in the trace MAY differ from the
sequence number reported on in any accompanying SR or RR report.
Note that because of sequence number wraparound, the ending sequence
number MAY be less than the beginning sequence number. A Loss RLE
Report Block MUST NOT be used to report upon a range of 65,534 or
greater in the sequence number space, as there is no means of
identifying multiple wraparounds.
The trace described by a Loss RLE report consists of a sequence of
Boolean values, one for each sequence number of the trace. A value
of one represents a packet receipt, meaning that one or more packets
having that sequence number have been received since the most recent
wraparound of sequence numbers (or since the beginning of the RTP
session if no wraparound has been judged to have occurred). A value
of zero represents a packet loss, meaning that there has been no
packet receipt for that sequence number as of the time of the report.
If a packet with a given sequence number is received after a report
of a loss for that sequence number, a later Loss RLE report MAY
report a packet receipt for that sequence number.
The encoding itself consists of a series of 16 bit units called
chunks that describe sequences of packet receipts or losses in the
trace. Each chunk either specifies a run length or a bit vector, or
is a null chunk. A run length describes between 1 and 16,383 events
that are all the same (either all receipts or all losses). A bit
vector describes 15 events that may be mixed receipts and losses. A
null chunk describes no events, and is used to round out the block to
a 32 bit word boundary.
The mapping from a sequence of lost and received packets into a
sequence of chunks is not necessarily unique. For example, the
following trace covers 45 packets, of which the 22nd and 24th have
been lost and the others received:
1111 1111 1111 1111 1111 1010 1111 1111 1111 1111 1111 1
Friedman, et al. Standards Track [Page 11]

RFC 3611 RTCP XR November 2003
One way to encode this would be:
bit vector 1111 1111 1111 111
bit vector 1111 1101 0111 111
bit vector 1111 1111 1111 111
null chunk
Another way to encode this would be:
run of 21 receipts
bit vector 0101 1111 1111 111
run of 9 receipts
null chunk
The choice of encoding is left to the application. As part of this
freedom of choice, applications MAY terminate a series of run length
and bit vector chunks with a bit vector chunk that runs beyond the
sequence number space described by the report block. For example, if
the 44th packet in the same sequence was lost:
1111 1111 1111 1111 1111 1010 1111 1111 1111 1111 1110 1
This could be encoded as:
run of 21 receipts
bit vector 0101 1111 1111 111
bit vector 1111 1110 1000 000
null chunk
In this example, the last five bits of the second bit vector describe
a part of the sequence number space that extends beyond the last
sequence number in the trace. These bits have been set to zero.
All bits in a bit vector chunk that describe a part of the sequence
number space that extends beyond the last sequence number in the
trace MUST be set to zero, and MUST be ignored by the receiver.
A null packet MUST appear at the end of a Loss RLE Report Block if
the number of run length plus bit vector chunks is odd. The null
chunk MUST NOT appear in any other context.
Caution should be used in sending Loss RLE Report Blocks because,
even with the compression provided by run length encoding, they can
easily consume bandwidth out of proportion with normal RTCP packets.
The block type includes a mechanism, called thinning, that allows an
application to limit report sizes.
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RFC 3611 RTCP XR November 2003
A thinning value, T, selects a subset of packets within the sequence
number space: those with sequence numbers that are multiples of 2^T.
Packet reception and loss reports apply only to those packets. T can
vary between 0 and 15. If T is zero, then every packet in the
sequence number space is reported upon. If T is fifteen, then one in
every 32,768 packets is reported upon.
Suppose that the trace just described begins at sequence number
13,821. The last sequence number in the trace is 13,865. If the
trace were to be thinned with a thinning value of T=2, then the
following sequence numbers would be reported upon: 13,824, 13,828,
13,832, 13,836, 13,840, 13,844, 13,848, 13,852, 13,856, 13,860,
13,864. The thinned trace would be as follows:
1 1 1 1 1 0 1 1 1 1 0
This could be encoded as follows:
bit vector 1111 1011 1100 000
null chunk
The last four bits in the bit vector, representing sequence numbers
13,868, 13,872, 13,876, and 13,880, extend beyond the trace and are
thus set to zero and are ignored by the receiver. With thinning, the
loss of the 22nd packet goes unreported because its sequence number,
13,842, is not a multiple of four. Packet receipts for all sequence
numbers that are not multiples of four also go unreported. However,
in this example thinning has permitted the Loss RLE Report Block to
be shortened by one 32 bit word.
Choice of the thinning value is left to the application.
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RFC 3611 RTCP XR November 2003
The Loss RLE Report Block has the following format:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| BT=1 | rsvd. | T | block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of source |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| begin_seq | end_seq |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| chunk 1 | chunk 2 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| chunk n-1 | chunk n |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
block type (BT): 8 bits
A Loss RLE Report Block is identified by the constant 1.
rsvd.: 4 bits
This field is reserved for future definition. In the absence
of such definition, the bits in this field MUST be set to zero
and MUST be ignored by the receiver.
thinning (T): 4 bits
The amount of thinning performed on the sequence number space.
Only those packets with sequence numbers 0 mod 2^T are reported
on by this block. A value of 0 indicates that there is no
thinning, and all packets are reported on. The maximum
thinning is one packet in every 32,768 (amounting to two
packets within each 16-bit sequence space).
block length: 16 bits
Defined in Section 3.
SSRC of source: 32 bits
The SSRC of the RTP data packet source being reported upon by
this report block.
begin_seq: 16 bits
The first sequence number that this block reports on.
end_seq: 16 bits
The last sequence number that this block reports on plus one.
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RFC 3611 RTCP XR November 2003
chunk i: 16 bits
There are three chunk types: run length, bit vector, and
terminating null, defined in the following sections. If the
chunk is all zeroes, then it is a terminating null chunk.
Otherwise, the left most bit of the chunk determines its type:
0 for run length and 1 for bit vector.
4.1.1. Run Length Chunk
0 1
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|C|R| run length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
chunk type (C): 1 bit
A zero identifies this as a run length chunk.
run type (R): 1 bit
Zero indicates a run of 0s. One indicates a run of 1s.
run length: 14 bits
A value between 1 and 16,383. The value MUST not be zero for a
run length chunk (zeroes in both the run type and run length
fields would make the chunk a terminating null chunk). Run
lengths of 15 or less MAY be described with a run length chunk
despite the fact that they could also be described as part of a
bit vector chunk.
4.1.2. Bit Vector Chunk
0 1
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|C| bit vector |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
chunk type (C): 1 bit
A one identifies this as a bit vector chunk.
bit vector: 15 bits
The vector is read from left to right, in order of increasing
sequence number (with the appropriate allowance for
wraparound).
Friedman, et al. Standards Track [Page 15]

RFC 3611 RTCP XR November 20034.1.3. Terminating Null Chunk
This chunk is all zeroes.
0 1
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
4.2. Duplicate RLE Report Block
This block type permits per-sequence-number reports on duplicates in
a source's RTP packet stream. Such information can be used for
network diagnosis, and provide an alternative to packet losses as a
basis for multicast tree topology inference.
The Duplicate RLE Report Block format is identical to the Loss RLE
Report Block format. Only the interpretation is different, in that
the information concerns packet duplicates rather than packet losses.
The trace to be encoded in this case also consists of zeros and ones,
but a zero here indicates the presence of duplicate packets for a
given sequence number, whereas a one indicates that no duplicates
were received.
The existence of a duplicate for a given sequence number is
determined over the entire reporting period. For example, if packet
number 12,593 arrives, followed by other packets with other sequence
numbers, the arrival later in the reporting period of another packet
numbered 12,593 counts as a duplicate for that sequence number. The
duplicate does not need to follow immediately upon the first packet
of that number. Care must be taken that a report does not cover a
range of 65,534 or greater in the sequence number space.
No distinction is made between the existence of a single duplicate
packet and multiple duplicate packets for a given sequence number.
Note also that since there is no duplicate for a lost packet, a loss
is encoded as a one in a Duplicate RLE Report Block.
Friedman, et al. Standards Track [Page 16]

RFC 3611 RTCP XR November 20034.3. Packet Receipt Times Report Block
This block type permits per-sequence-number reports on packet receipt
times for a given source's RTP packet stream. Such information can
be used for MINC inference of the topology of the multicast tree used
to distribute the source's RTP packets, and of the delays along the
links within that tree. It can also be used to measure partial path
characteristics and to model distributions for packet jitter.
Packet receipt times are expressed in the same units as in the RTP
timestamps of data packets. This is so that, for each packet, one
can establish both the send time and the receipt time in comparable
terms. Note, however, that as an RTP sender ordinarily initializes
its time to a value chosen at random, there can be no expectation
that reported send and receipt times will differ by an amount equal
to the one-way delay between sender and receiver. The reported times
can nonetheless be useful for the purposes mentioned above.
At least one packet MUST have been received for each sequence number
reported upon in this block. If this block type is used to report
receipt times for a series of sequence numbers that includes lost
packets, several blocks are required. If duplicate packets have been
received for a given sequence number, and those packets differ in
their receipt times, any time other than the earliest MUST NOT be
reported. This is to ensure consistency among reports.
Times reported in RTP timestamp format consume more bits than loss or
duplicate information, and do not lend themselves to run length
encoding. The use of thinning is encouraged to limit the size of
Packet Receipt Times Report Blocks.
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RFC 3611 RTCP XR November 2003
Packet i receipt time: 32 bits
The receipt time of the packet with sequence number i at the
receiver. The modular arithmetic shown in the packet format
diagram is to allow for sequence number rollover. It is
preferable for the time value to be established at the link
layer interface, or in any case as close as possible to the
wire arrival time. Units and format are the same as for the
timestamp in RTP data packets. As opposed to RTP data packet
timestamps, in which nominal values may be used instead of
system clock values in order to convey information useful for
periodic playout, the receipt times should reflect the actual
time as closely as possible. For a session, if the RTP
timestamp is chosen at random, the first receipt time value
SHOULD also be chosen at random, and subsequent timestamps
offset from this value. On the other hand, if the RTP
timestamp is meant to reflect the reference time at the sender,
then the receipt time SHOULD be as close as possible to the
reference time at the receiver.
4.4. Receiver Reference Time Report Block
This block extends RTCP's timestamp reporting so that non-senders may
also send timestamps. It recapitulates the NTP timestamp fields from
the RTCP Sender Report [9, Sec. 6.3.1]. A non-sender may estimate
its round trip time (RTT) to other participants, as proposed in [18],
by sending this report block and receiving DLRR Report Blocks (see
next section) in reply.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| BT=4 | reserved | block length = 2 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| NTP timestamp, most significant word |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| NTP timestamp, least significant word |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
block type (BT): 8 bits
A Receiver Reference Time Report Block is identified by the
constant 4.
reserved: 8 bits
This field is reserved for future definition. In the absence
of such definition, the bits in this field MUST be set to zero
and MUST be ignored by the receiver.
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RFC 3611 RTCP XR November 2003
block length: 16 bits
The constant 2, in accordance with the definition of this field
in Section 3.
NTP timestamp: 64 bits
Indicates the wallclock time when this block was sent so that
it may be used in combination with timestamps returned in DLRR
Report Blocks (see next section) from other receivers to
measure round-trip propagation to those receivers. Receivers
should expect that the measurement accuracy of the timestamp
may be limited to far less than the resolution of the NTP
timestamp. The measurement uncertainty of the timestamp is not
indicated as it may not be known. A report block sender that
can keep track of elapsed time but has no notion of wallclock
time may use the elapsed time since joining the session
instead. This is assumed to be less than 68 years, so the high
bit will be zero. It is permissible to use the sampling clock
to estimate elapsed wallclock time. A report sender that has
no notion of wallclock or elapsed time may set the NTP
timestamp to zero.
4.5. DLRR Report Block
This block extends RTCP's delay since the last Sender Report (DLSR)
mechanism [9, Sec. 6.3.1] so that non-senders may also calculate
round trip times, as proposed in [18]. It is termed DLRR for delay
since the last Receiver Report, and may be sent in response to a
Receiver Timestamp Report Block (see previous section) from a
receiver to allow that receiver to calculate its round trip time to
the respondent. The report consists of one or more 3 word sub-
blocks: one sub-block per Receiver Report.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| BT=5 | reserved | block length |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| SSRC_1 (SSRC of first receiver) | sub-
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block
| last RR (LRR) | 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| delay since last RR (DLRR) |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| SSRC_2 (SSRC of second receiver) | sub-
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block
: ... : 2
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
Friedman, et al. Standards Track [Page 21]

RFC 3611 RTCP XR November 2003
block type (BT): 8 bits
A DLRR Report Block is identified by the constant 5.
reserved: 8 bits
This field is reserved for future definition. In the absence
of such definition, the bits in this field MUST be set to zero
and MUST be ignored by the receiver.
block length: 16 bits
Defined in Section 3.
last RR timestamp (LRR): 32 bits
The middle 32 bits out of 64 in the NTP timestamp (as explained
in the previous section), received as part of a Receiver
Reference Time Report Block from participant SSRC_n. If no
such block has been received, the field is set to zero.
delay since last RR (DLRR): 32 bits
The delay, expressed in units of 1/65536 seconds, between
receiving the last Receiver Reference Time Report Block from
participant SSRC_n and sending this DLRR Report Block. If a
Receiver Reference Time Report Block has yet to be received
from SSRC_n, the DLRR field is set to zero (or the DLRR is
omitted entirely). Let SSRC_r denote the receiver issuing this
DLRR Report Block. Participant SSRC_n can compute the round-
trip propagation delay to SSRC_r by recording the time A when
this Receiver Timestamp Report Block is received. It
calculates the total round-trip time A-LRR using the last RR
timestamp (LRR) field, and then subtracting this field to leave
the round-trip propagation delay as A-LRR-DLRR. This is
illustrated in [9, Fig. 2].
4.6. Statistics Summary Report Block
This block reports statistics beyond the information carried in the
standard RTCP packet format, but is not as finely grained as that
carried in the report blocks previously described. Information is
recorded about lost packets, duplicate packets, jitter measurements,
and TTL or Hop Limit values. Such information can be useful for
network management.
The report block contents are dependent upon a series of flag bits
carried in the first part of the header. Not all parameters need to
be reported in each block. Flags indicate which are and which are
not reported. The fields corresponding to unreported parameters MUST
be present, but are set to zero. The receiver MUST ignore any
Statistics Summary Report Block with a non-zero value in any field
flagged as unreported.
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RFC 3611 RTCP XR November 2003
IPv6 Hop Limit values. The value 3 is undefined and MUST NOT
be used.
rsvd.: 3 bits
This field is reserved for future definition. In the absence
of such a definition, the bits in this field MUST be set to
zero and MUST be ignored by the receiver.
block length: 16 bits
The constant 9, in accordance with the definition of this field
in Section 3.
SSRC of source: 32 bits
As defined in Section 4.1.
begin_seq: 16 bits
As defined in Section 4.1.
end_seq: 16 bits
As defined in Section 4.1.
lost_packets: 32 bits
Number of lost packets in the above sequence number interval.
dup_packets: 32 bits
Number of duplicate packets in the above sequence number
interval.
min_jitter: 32 bits
The minimum relative transit time between two packets in the
above sequence number interval. All jitter values are measured
as the difference between a packet's RTP timestamp and the
reporter's clock at the time of arrival, measured in the same
units.
max_jitter: 32 bits
The maximum relative transit time between two packets in the
above sequence number interval.
mean_jitter: 32 bits
The mean relative transit time between each two packet series
in the above sequence number interval, rounded to the nearest
value expressible as an RTP timestamp.
dev_jitter: 32 bits
The standard deviation of the relative transit time between
each two packet series in the above sequence number interval.
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RFC 3611 RTCP XR November 2003
min_ttl_or_hl: 8 bits
The minimum TTL or Hop Limit value of data packets in the
sequence number range.
max_ttl_or_hl: 8 bits
The maximum TTL or Hop Limit value of data packets in the
sequence number range.
mean_ttl_or_hl: 8 bits
The mean TTL or Hop Limit value of data packets in the sequence
number range, rounded to the nearest integer.
dev_ttl_or_hl: 8 bits
The standard deviation of TTL or Hop Limit values of data
packets in the sequence number range.
4.7. VoIP Metrics Report Block
The VoIP Metrics Report Block provides metrics for monitoring voice
over IP (VoIP) calls. These metrics include packet loss and discard
metrics, delay metrics, analog metrics, and voice quality metrics.
The block reports separately on packets lost on the IP channel, and
those that have been received but then discarded by the receiving
jitter buffer. It also reports on the combined effect of losses and
discards, as both have equal effect on call quality.
In order to properly assess the quality of a Voice over IP call, it
is desirable to consider the degree of burstiness of packet loss
[14]. Following a Gilbert-Elliott model [3], a period of time,
bounded by lost and/or discarded packets with a high rate of losses
and/or discards, is a "burst", and a period of time between two
bursts is a "gap". Bursts correspond to periods of time during which
the packet loss rate is high enough to produce noticeable degradation
in audio quality. Gaps correspond to periods of time during which
only isolated lost packets may occur, and in general these can be
masked by packet loss concealment. Delay reports include the transit
delay between RTP end points and the VoIP end system processing
delays, both of which contribute to the user perceived delay.
Additional metrics include signal, echo, noise, and distortion
levels. Call quality metrics include R factors (as described by the
E Model defined in [6,3]) and mean opinion scores (MOS scores).
Implementations MUST provide values for all the fields defined here.
For certain metrics, if the value is undefined or unknown, then the
specified default or unknown field value MUST be provided.
Friedman, et al. Standards Track [Page 25]

RFC 3611 RTCP XR November 20034.7.1. Packet Loss and Discard Metrics
It is very useful to distinguish between packets lost by the network
and those discarded due to jitter. Both have equal effect on the
quality of the voice stream, however, having separate counts helps
identify the source of quality degradation. These fields MUST be
populated, and MUST be set to zero if no packets have been received.
loss rate: 8 bits
The fraction of RTP data packets from the source lost since the
beginning of reception, expressed as a fixed point number with
the binary point at the left edge of the field. This value is
calculated by dividing the total number of packets lost (after
the effects of applying any error protection such as FEC) by
the total number of packets expected, multiplying the result of
the division by 256, limiting the maximum value to 255 (to
avoid overflow), and taking the integer part. The numbers of
duplicated packets and discarded packets do not enter into this
calculation. Since receivers cannot be required to maintain
unlimited buffers, a receiver MAY categorize late-arriving
packets as lost. The degree of lateness that triggers a loss
SHOULD be significantly greater than that which triggers a
discard.
discard rate: 8 bits
The fraction of RTP data packets from the source that have been
discarded since the beginning of reception, due to late or
early arrival, under-run or overflow at the receiving jitter
buffer. This value is expressed as a fixed point number with
the binary point at the left edge of the field. It is
calculated by dividing the total number of packets discarded
(excluding duplicate packet discards) by the total number of
packets expected, multiplying the result of the division by
256, limiting the maximum value to 255 (to avoid overflow), and
taking the integer part.
4.7.2. Burst Metrics
A burst is a period during which a high proportion of packets are
either lost or discarded due to late arrival. A burst is defined, in
terms of a value Gmin, as the longest sequence that (a) starts with a
lost or discarded packet, (b) does not contain any occurrences of
Gmin or more consecutively received (and not discarded) packets, and
(c) ends with a lost or discarded packet.
A gap, informally, is a period of low packet losses and/or discards.
Formally, a gap is defined as any of the following: (a) the period
from the start of an RTP session to the receipt time of the last
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RFC 3611 RTCP XR November 2003
received packet before the first burst, (b) the period from the end
of the last burst to either the time of the report or the end of the
RTP session, whichever comes first, or (c) the period of time between
two bursts.
For the purpose of determining if a lost or discarded packet near the
start or end of an RTP session is within a gap or a burst, it is
assumed that the RTP session is preceded and followed by at least
Gmin received packets, and that the time of the report is followed by
at least Gmin received packets.
A gap has the property that any lost or discarded packets within the
gap must be preceded and followed by at least Gmin packets that were
received and not discarded. This gives a maximum loss/discard rate
within a gap of: 1 / (Gmin + 1).
A Gmin value of 16 is RECOMMENDED, as it results in gap
characteristics that correspond to good quality (i.e., low packet
loss rate, a minimum distance of 16 received packets between lost
packets), and hence differentiates nicely between good and poor
quality periods.
For example, a 1 denotes a received packet, 0 a lost packet, and X a
discarded packet in the following pattern covering 64 packets:
11110111111111111111111X111X1011110111111111111111111X111111111
|---------gap----------|--burst---|------------gap------------|
The burst consists of the twelve packets indicated above, starting at
a discarded packet and ending at a lost packet. The first gap starts
at the beginning of the session and the second gap ends at the time
of the report.
If the packet spacing is 10 ms and the Gmin value is the recommended
value of 16, the burst duration is 120 ms, the burst density 0.33,
the gap duration 230 ms + 290 ms = 520 ms, and the gap density 0.04.
This would result in reported values as follows (see field
descriptions for semantics and details on how these are calculated):
loss rate 12, which corresponds to 5%
discard rate 12, which corresponds to 5%
burst density 84, which corresponds to 33%
gap density 10, which corresponds to 4%
burst duration 120, value in milliseconds
gap duration 520, value in milliseconds
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RFC 3611 RTCP XR November 2003
burst density: 8 bits
The fraction of RTP data packets within burst periods since the
beginning of reception that were either lost or discarded.
This value is expressed as a fixed point number with the binary
point at the left edge of the field. It is calculated by
dividing the total number of packets lost or discarded
(excluding duplicate packet discards) within burst periods by
the total number of packets expected within the burst periods,
multiplying the result of the division by 256, limiting the
maximum value to 255 (to avoid overflow), and taking the
integer part. This field MUST be populated and MUST be set to
zero if no packets have been received.
gap density: 8 bits
The fraction of RTP data packets within inter-burst gaps since
the beginning of reception that were either lost or discarded.
The value is expressed as a fixed point number with the binary
point at the left edge of the field. It is calculated by
dividing the total number of packets lost or discarded
(excluding duplicate packet discards) within gap periods by the
total number of packets expected within the gap periods,
multiplying the result of the division by 256, limiting the
maximum value to 255 (to avoid overflow), and taking the
integer part. This field MUST be populated and MUST be set to
zero if no packets have been received.
burst duration: 16 bits
The mean duration, expressed in milliseconds, of the burst
periods that have occurred since the beginning of reception.
The duration of each period is calculated based upon the
packets that mark the beginning and end of that period. It is
equal to the timestamp of the end packet, plus the duration of
the end packet, minus the timestamp of the beginning packet.
If the actual values are not available, estimated values MUST
be used. If there have been no burst periods, the burst
duration value MUST be zero.
gap duration: 16 bits
The mean duration, expressed in milliseconds, of the gap
periods that have occurred since the beginning of reception.
The duration of each period is calculated based upon the packet
that marks the end of the prior burst and the packet that marks
the beginning of the subsequent burst. It is equal to the
timestamp of the subsequent burst packet, minus the timestamp
of the prior burst packet, plus the duration of the prior burst
packet. If the actual values are not available, estimated
values MUST be used. In the case of a gap that occurs at the
beginning of reception, the sum of the timestamp of the prior
Friedman, et al. Standards Track [Page 29]

RFC 3611 RTCP XR November 2003
burst packet and the duration of the prior burst packet are
replaced by the reception start time. In the case of a gap
that occurs at the end of reception, the timestamp of the
subsequent burst packet is replaced by the reception end time.
If there have been no gap periods, the gap duration value MUST
be zero.
4.7.3. Delay Metrics
For the purpose of the following definitions, the RTP interface is
the interface between the RTP instance and the voice application
(i.e., FEC, de-interleaving, de-multiplexing, jitter buffer). For
example, the time delay due to RTP payload multiplexing would be
considered part of the voice application or end-system delay, whereas
delay due to multiplexing RTP frames within a UDP frame would be
considered part of the RTP reported delay. This distinction is
consistent with the use of RTCP for delay measurements.
round trip delay: 16 bits
The most recently calculated round trip time between RTP
interfaces, expressed in milliseconds. This value MAY be
measured using RTCP, the DLRR method defined in Section 4.5 of
this document, where it is necessary to convert the units of
measurement from NTP timestamp values to milliseconds, or other
approaches. If RTCP is used, then the reported delay value is
the time of receipt of the most recent RTCP packet from source
SSRC, minus the LSR (last SR) time reported in its SR (Sender
Report), minus the DLSR (delay since last SR) reported in its
SR. A non-zero LSR value is required in order to calculate
round trip delay. A value of 0 is permissible; however, this
field MUST be populated as soon as a delay estimate is
available.
end system delay: 16 bits
The most recently estimated end system delay, expressed in
milliseconds. End system delay is defined as the sum of the
total sample accumulation and encoding delay associated with
the sending direction and the jitter buffer, decoding, and
playout buffer delay associated with the receiving direction.
This delay MAY be estimated or measured. This value SHOULD be
provided in all VoIP metrics reports. If an implementation is
unable to provide the data, the value 0 MUST be used.
Friedman, et al. Standards Track [Page 30]

RFC 3611 RTCP XR November 2003
Note that the one way symmetric VoIP segment delay may be calculated
from the round trip and end system delays is as follows; if the round
trip delay is denoted, RTD and the end system delays associated with
the two endpoints are ESD(A) and ESD(B) then:
one way symmetric voice path delay = ( RTD + ESD(A) + ESD(B) ) / 2
4.7.4. Signal Related Metrics
The following metrics are intended to provide real time information
related to the non-packet elements of the voice over IP system to
assist with the identification of problems affecting call quality.
The values identified below must be determined for the received audio
signal. The information required to populate these fields may not be
available in all systems, although it is strongly recommended that
this data SHOULD be provided to support problem diagnosis.
signal level: 8 bits
The voice signal relative level is defined as the ratio of the
signal level to a 0 dBm0 reference [10], expressed in decibels
as a signed integer in two's complement form. This is measured
only for packets containing speech energy. The intent of this
metric is not to provide a precise measurement of the signal
level but to provide a real time indication that the signal
level may be excessively high or low.
signal level = 10 Log10 ( rms talkspurt power (mW) )
A value of 127 indicates that this parameter is unavailable.
Typical values should generally be in the -15 to -20 dBm range.
noise level: 8 bits
The noise level is defined as the ratio of the silent period
background noise level to a 0 dBm0 reference, expressed in
decibels as a signed integer in two's complement form.
noise level = 10 Log10 ( rms silence power (mW) )
A value of 127 indicates that this parameter is unavailable.
residual echo return loss (RERL): 8 bits
The residual echo return loss value may be measured directly by
the VoIP end system's echo canceller or may be estimated by
adding the echo return loss (ERL) and echo return loss
enhancement (ERLE) values reported by the echo canceller.
RERL(dB) = ERL (dB) + ERLE (dB)
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RFC 3611 RTCP XR November 2003
In the case of a VoIP gateway, the source of echo is typically
line echo that occurs at 2-4 wire conversion points in the
network. This can be in the 8-12 dB range. A line echo
canceler can provide an ERLE of 30 dB or more and hence reduce
this to 40-50 dB. In the case of an IP phone, this could be
acoustic coupling between handset speaker and microphone or
residual acoustic echo from speakerphone operation, and may
more correctly be termed terminal coupling loss (TCL). A
typical handset would result in 40-50 dB of echo loss due to
acoustic feedback.
Examples:
- IP gateway connected to circuit switched network with 2 wire
loop. Without echo cancellation, typical 2-4 wire converter
ERL of 12 dB. RERL = ERL + ERLE = 12 + 0 = 12 dB.
- IP gateway connected to circuit switched network with 2 wire
loop. With echo canceler that improves echo by 30 dB.
RERL = ERL + ERLE = 12 + 30 = 42 dB.
- IP phone with conventional handset. Acoustic coupling from
handset speaker to microphone (terminal coupling loss) is
typically 40 dB. RERL = TCL = 40 dB.
If we denote the local end of the VoIP path as A and the remote
end as B, and if the sender loudness rating (SLR) and receiver
loudness rating (RLR) are known for A (default values 8 dB and
2 dB respectively), then the echo loudness level at end A
(talker echo loudness rating or TELR) is given by:
TELR(A) = SRL(A) + ERL(B) + ERLE(B) + RLR(A)
TELR(B) = SRL(B) + ERL(A) + ERLE(A) + RLR(B)
Hence, in order to incorporate echo into a voice quality
estimate at the A end of a VoIP connection, it is desirable to
send the ERL + ERLE value from B to A using a format such as
RTCP XR.
Echo related information may not be available in all VoIP end
systems. As echo does have a significant effect on
conversational quality, it is recommended that estimated values
for echo return loss and terminal coupling loss be provided (if
sensible estimates can be reasonably determined).
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RFC 3611 RTCP XR November 2003
Typical values for end systems are given below to provide
guidance:
- IP Phone with handset: typically 45 dB.
- PC softphone or speakerphone: extremely variable, consider
reporting "undefined" (127).
- IP gateway with line echo canceller: typically has ERL and
ERLE available.
- IP gateway without line echo canceller: frequently a source
of echo related problems, consider reporting either a low
value (12 dB) or "undefined" (127).
Gmin
See Configuration Parameters (Section 4.7.6, below).
4.7.5. Call Quality or Transmission Quality Metrics
The following metrics are direct measures of the call quality or
transmission quality, and incorporate the effects of codec type,
packet loss, discard, burstiness, delay etc. These metrics may not
be available in all systems, however, they SHOULD be provided in
order to support problem diagnosis.
R factor: 8 bits
The R factor is a voice quality metric describing the segment
of the call that is carried over this RTP session. It is
expressed as an integer in the range 0 to 100, with a value of
94 corresponding to "toll quality" and values of 50 or less
regarded as unusable. This metric is defined as including the
effects of delay, consistent with ITU-T G.107 [6] and ETSI TS
101 329-5 [3].
A value of 127 indicates that this parameter is unavailable.
Values other than 127 and the valid range defined above MUST
not be sent and MUST be ignored by the receiving system.
ext. R factor: 8 bits
The external R factor is a voice quality metric describing the
segment of the call that is carried over a network segment
external to the RTP segment, for example a cellular network.
Its values are interpreted in the same manner as for the RTP R
factor. This metric is defined as including the effects of
delay, consistent with ITU-T G.107 [6] and ETSI TS 101 329-5
[3], and relates to the outward voice path from the Voice over
IP termination for which this metrics block applies.
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RFC 3611 RTCP XR November 2003
A value of 127 indicates that this parameter is unavailable.
Values other than 127 and the valid range defined above MUST
not be sent and MUST be ignored by the receiving system.
Note that an overall R factor may be estimated from the RTP segment R
factor and the external R factor, as follows:
R total = RTP R factor + ext. R factor - 94
MOS-LQ: 8 bits
The estimated mean opinion score for listening quality (MOS-LQ)
is a voice quality metric on a scale from 1 to 5, in which 5
represents excellent and 1 represents unacceptable. This
metric is defined as not including the effects of delay and can
be compared to MOS scores obtained from listening quality (ACR)
tests. It is expressed as an integer in the range 10 to 50,
corresponding to MOS x 10. For example, a value of 35 would
correspond to an estimated MOS score of 3.5.
A value of 127 indicates that this parameter is unavailable.
Values other than 127 and the valid range defined above MUST
not be sent and MUST be ignored by the receiving system.
MOS-CQ: 8 bits
The estimated mean opinion score for conversational quality
(MOS-CQ) is defined as including the effects of delay and other
effects that would affect conversational quality. The metric
may be calculated by converting an R factor determined
according to ITU-T G.107 [6] or ETSI TS 101 329-5 [3] into an
estimated MOS using the equation specified in G.107. It is
expressed as an integer in the range 10 to 50, corresponding to
MOS x 10, as for MOS-LQ.
A value of 127 indicates that this parameter is unavailable.
Values other than 127 and the valid range defined above MUST
not be sent and MUST be ignored by the receiving system.
4.7.6. Configuration Parameters
Gmin: 8 bits
The gap threshold. This field contains the value used for this
report block to determine if a gap exists. The recommended
value of 16 corresponds to a burst period having a minimum
density of 6.25% of lost or discarded packets, which may cause
noticeable degradation in call quality; during gap periods, if
packet loss or discard occurs, each lost or discarded packet
would be preceded by and followed by a sequence of at least 16
received non-discarded packets. Note that lost or discarded
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RFC 3611 RTCP XR November 2003
packets that occur within Gmin packets of a report being
generated may be reclassified as part of a burst or gap in
later reports. ETSI TS 101 329-5 [3] defines a computationally
efficient algorithm for measuring burst and gap density using a
packet loss/discard event driven approach. This algorithm is
reproduced in Appendix A.2 of the present document. Gmin MUST
not be zero, MUST be provided, and MUST remain constant across
VoIP Metrics report blocks for the duration of the RTP session.
receiver configuration byte (RX config): 8 bits
This byte consists of the following fields:
0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+
|PLC|JBA|JB rate|
+-+-+-+-+-+-+-+-+
packet loss concealment (PLC): 2 bits
Standard (11) / enhanced (10) / disabled (01) / unspecified
(00). When PLC = 11, then a simple replay or interpolation
algorithm is being used to fill-in the missing packet; this
approach is typically able to conceal isolated lost packets at
low packet loss rates. When PLC = 10, then an enhanced
interpolation algorithm is being used; algorithms of this type
are able to conceal high packet loss rates effectively. When
PLC = 01, then silence is being inserted in place of lost
packets. When PLC = 00, then no information is available
concerning the use of PLC; however, for some codecs this may be
inferred.
jitter buffer adaptive (JBA): 2 bits
Adaptive (11) / non-adaptive (10) / reserved (01)/ unknown
(00). When the jitter buffer is adaptive, then its size is
being dynamically adjusted to deal with varying levels of
jitter. When non-adaptive, the jitter buffer size is
maintained at a fixed level. When either adaptive or non-
adaptive modes are specified, then the jitter buffer size
parameters below MUST be specified.
jitter buffer rate (JB rate): 4 bits
J = adjustment rate (0-15). This represents the implementation
specific adjustment rate of a jitter buffer in adaptive mode.
This parameter is defined in terms of the approximate time
taken to fully adjust to a step change in peak to peak jitter
from 30 ms to 100 ms such that:
adjustment time = 2 * J * frame size (ms)
Friedman, et al. Standards Track [Page 35]

RFC 3611 RTCP XR November 2003
This parameter is intended only to provide a guide to the
degree of "aggressiveness" of an adaptive jitter buffer and may
be estimated. A value of 0 indicates that the adjustment time
is unknown for this implementation.
reserved: 8 bits
This field is reserved for future definition. In the absence
of such a definition, the bits in this field MUST be set to
zero and MUST be ignored by the receiver.
4.7.7. Jitter Buffer Parameters
The values reported in these fields SHOULD be the most recently
obtained values at the time of reporting.
jitter buffer nominal delay (JB nominal): 16 bits
This is the current nominal jitter buffer delay in
milliseconds, which corresponds to the nominal jitter buffer
delay for packets that arrive exactly on time. This parameter
MUST be provided for both fixed and adaptive jitter buffer
implementations.
jitter buffer maximum delay (JB maximum): 16 bits
This is the current maximum jitter buffer delay in milliseconds
which corresponds to the earliest arriving packet that would
not be discarded. In simple queue implementations this may
correspond to the nominal size. In adaptive jitter buffer
implementations, this value may dynamically vary up to JB abs
max (see below). This parameter MUST be provided for both
fixed and adaptive jitter buffer implementations.
jitter buffer absolute maximum delay (JB abs max): 16 bits
This is the absolute maximum delay in milliseconds that the
adaptive jitter buffer can reach under worst case conditions.
If this value exceeds 65535 milliseconds, then this field SHALL
convey the value 65535. This parameter MUST be provided for
adaptive jitter buffer implementations and its value MUST be
set to JB maximum for fixed jitter buffer implementations.
5. SDP Signaling
This section defines Session Description Protocol (SDP) [4] signaling
for XR blocks that can be employed by applications that utilize SDP.
This signaling is defined to be used either by applications that
implement the SDP Offer/Answer model [8] or by applications that use
SDP to describe media and transport configurations in connection
Friedman, et al. Standards Track [Page 36]

RFC 3611 RTCP XR November 2003
with such protocols as the Session Announcement Protocol (SAP) [15]
or the Real Time Streaming Protocol (RTSP) [17]. There exist other
potential signaling methods that are not defined here.
The XR blocks MAY be used without prior signaling. This is
consistent with the rules governing other RTCP packet types, as
described in [9]. An example in which signaling would not be used is
an application that always requires the use of one or more XR blocks.
However, for applications that are configured at session initiation,
the use of some type of signaling is recommended.
Note that, although the use of SDP signaling for XR blocks may be
optional, if used, it MUST be used as defined here. If SDP signaling
is used in an environment where XR blocks are only implemented by
some fraction of the participants, the ones not implementing the XR
blocks will ignore the SDP attribute.
5.1. The SDP Attribute
This section defines one new SDP attribute "rtcp-xr" that can be used
to signal participants in a media session that they should use the
specified XR blocks. This attribute can be easily extended in the
future with new parameters to cover any new report blocks.
The RTCP XR blocks SDP attribute is defined below in Augmented
Backus-Naur Form (ABNF) [2]. It is both a session and a media level
attribute. When specified at session level, it applies to all media
level blocks in the session. Any media level specification MUST
replace a session level specification, if one is present, for that
media block.
rtcp-xr-attrib = "a=" "rtcp-xr" ":" [xr-format *(SP xr-format)] CRLF
xr-format = pkt-loss-rle
/ pkt-dup-rle
/ pkt-rcpt-times
/ rcvr-rtt
/ stat-summary
/ voip-metrics
/ format-ext
pkt-loss-rle = "pkt-loss-rle" ["=" max-size]
pkt-dup-rle = "pkt-dup-rle" ["=" max-size]
pkt-rcpt-times = "pkt-rcpt-times" ["=" max-size]
rcvr-rtt = "rcvr-rtt" "=" rcvr-rtt-mode [":" max-size]
rcvr-rtt-mode = "all"
/ "sender"
stat-summary = "stat-summary" ["=" stat-flag *("," stat-flag)]
Friedman, et al. Standards Track [Page 37]

RFC 3611 RTCP XR November 2003
stat-flag = "loss"
/ "dup"
/ "jitt"
/ "TTL"
/ "HL"
voip-metrics = "voip-metrics"
max-size = 1*DIGIT ; maximum block size in octets
DIGIT = %x30-39
format-ext = non-ws-string
non-ws-string = 1*(%x21-FF)
CRLF = %d13.10
The "rtcp-xr" attribute contains zero, one, or more XR block related
parameters. Each parameter signals functionality for an XR block, or
a group of XR blocks. The attribute is extensible so that parameters
can be defined for any future XR block (and a parameter should be
defined for every future block).
Each "rtcp-xr" parameter belongs to one of two categories. The first
category, the unilateral parameters, are for report blocks that
simply report on the RTP stream and related metrics. The second
category, collaborative parameters, are for XR blocks that involve
actions by more than one party in order to carry out their functions.
Round trip time (RTT) measurement is an example of collaborative
functionality. An RTP data packet receiver sends a Receiver
Reference Time Report Block (Section 4.4). A participant that
receives this block sends a DLRR Report Block (Section 4.5) in
response, allowing the receiver to calculate its RTT to that
participant. As this example illustrates, collaborative
functionality may be implemented by two or more different XR blocks.
The collaborative functionality of several XR blocks may be governed
by a single "rtcp-xr" parameter.
For the unilateral category, this document defines the following
parameters. The parameter names and their corresponding XR formats
are as follows:
Parameter name XR block (block type and name)
-------------- ------------------------------------
pkt-loss-rle 1 Loss RLE Report Block
pkt-dup-rle 2 Duplicate RLE Report Block
pkt-rcpt-times 3 Packet Receipt Times Report Block
stat-summary 6 Statistics Summary Report Block
voip-metrics 7 VoIP Metrics Report Block
Friedman, et al. Standards Track [Page 38]

RFC 3611 RTCP XR November 2003
The "pkt-loss-rle", "pkt-dup-rle", and "pkt-rcpt-times" parameters
MAY specify an integer value. This value indicates the largest size
the whole report block SHOULD have in octets. This shall be seen as
an indication that thinning shall be applied if necessary to meet the
target size.
The "stat-summary" parameter contains a list indicating which fields
SHOULD be included in the Statistics Summary report blocks that are
sent. The list is a comma separated list, containing one or more
field indicators. The space character (0x20) SHALL NOT be present
within the list. Field indicators represent the flags defined in
Section 4.6. The field indicators and their respective flags are as
follows:
Indicator Flag
--------- ---------------------------
loss loss report flag (L)
dup duplicate report flag (D)
jitt jitter flag (J)
TTL TTL or Hop Limit flag (ToH)
HL TTL or Hop Limit flag (ToH)
For "loss", "dup", and "jitt", the presence of the indicator
indicates that the corresponding flag should be set to 1 in the
Statistics Summary report blocks that are sent. The presence of
"TTL" indicates that the corresponding flag should be set to 1. The
presence of "HL" indicates that the corresponding flag should be set
to 2. The indicators "TTL" and "HL" MUST NOT be signaled together.
Blocks in the collaborative category are classified as initiator
blocks or response blocks. Signaling SHOULD indicate which
participants are required to respond to the initiator block. A party
that wishes to receive response blocks from those participants can
trigger this by sending an initiator block.
The collaborative category currently consists only of one
functionality, namely the RTT measurement mechanism for RTP data
receivers. The collective functionality of the Receiver Reference
Time Report Block and DLRR Report Block is represented by the "rcvr-
rtt" parameter. This parameter takes as its arguments a mode value
and, optionally, a maximum size for the DLRR report block. The mode
value "all" indicates that both RTP data senders and data receivers
MAY send DLRR blocks, while the mode value "sender" indicates that
only active RTP senders MAY send DLRR blocks, i.e., non RTP senders
SHALL NOT send DLRR blocks. If a maximum size in octets is included,
any DLRR Report Blocks that are sent SHALL NOT exceed the specified
size. If size limitations mean that a DLRR Report Block sender
cannot report in one block upon all participants from which it has
Friedman, et al. Standards Track [Page 39]

RFC 3611 RTCP XR November 2003
received a Receiver Reference Time Report Block then it SHOULD report
on participants in a round robin fashion across several report
intervals.
The "rtcp-xr" attributes parameter list MAY be empty. This is useful
in cases in which an application needs to signal that it understands
the SDP signaling but does not wish to avail itself of XR
functionality. For example, an application in a SIP controlled
session could signal that it wishes to stop using all XR blocks by
removing all applicable SDP parameters in a re-INVITE message that it
sends. If XR blocks are not to be used at all from the beginning of
a session, it is RECOMMENDED that the "rtcp-xr" attribute not be
supplied at all.
When the "rtcp-xr" attribute is present, participants SHOULD NOT send
XR blocks other than the ones indicated by the parameters. This
means that inclusion of a "rtcp-xr" attribute without any parameters
tells a participant that it SHOULD NOT send any XR blocks at all.
The purpose is to conserve bandwidth. This is especially important
when collaborative parameters are applied to a large multicast group:
the sending of an initiator block could potentially trigger responses
from all participants. There are, however, contexts in which it
makes sense to send an XR block in the absence of a parameter
signaling its use. For instance, an application might be designed so
as to send certain report blocks without negotiation, while using SDP
signaling to negotiate the use of other blocks.
5.2. Usage in Offer/Answer
In the Offer/Answer context [8], the interpretation of SDP signaling
for XR packets depends upon the direction attribute that is signaled:
"recvonly", "sendrecv", or "sendonly" [4]. If no direction attribute
is supplied, then "sendrecv" is assumed. This section applies only
to unicast media streams, except where noted. Discussion of
unilateral parameters is followed by discussion of collaborative
parameters in this section.
For "sendonly" and "sendrecv" media stream offers that specify
unilateral "rtcp-xr" attribute parameters, the answerer SHOULD send
the corresponding XR blocks. For "sendrecv" offers, the answerer MAY
include the "rtcp-xr" attribute in its response, and specify any
unilateral parameters in order to request that the offerer send the
corresponding XR blocks. The offerer SHOULD send these blocks.
For "recvonly" media stream offers, the offerer's use of the "rtcp-
xr" attribute in connection with unilateral parameters indicates that
the offerer is capable of sending the corresponding XR blocks. If
Friedman, et al. Standards Track [Page 40]

RFC 3611 RTCP XR November 2003
the answerer responds with an "rtcp-xr" attribute, the offerer SHOULD
send XR blocks for each specified unilateral parameter that was in
its offer.
For multicast media streams, the inclusion of an "rtcp-xr" attribute
with unilateral parameters means that every media recipient SHOULD
send the corresponding XR blocks.
An SDP offer with a collaborative parameter declares the offerer
capable of receiving the corresponding initiator and replying with
the appropriate responses. For example, an offer that specifies the
"rcvr-rtt" parameter means that the offerer is prepared to receive
Receiver Reference Time Report Blocks and to send DLRR Report Blocks.
An offer of a collaborative parameter means that the answerer MAY
send the initiator, and, having received the initiator, the offerer
SHOULD send the responses.
There are exceptions to the rule that an offerer of a collaborative
parameter should send responses. For instance, the collaborative
parameter might specify a mode that excludes the offerer; or
congestion control or maximum transmission unit considerations might
militate against the offerer's response.
By including a collaborative parameter in its answer, the answerer
declares its ability to receive initiators and to send responses.
The offerer MAY then send initiators, to which the answerer SHOULD
reply with responses. As for the offer of a collaborative parameter,
there are exceptions to the rule that the answerer should reply.
When making an SDP offer of a collaborative parameter for a multicast
media stream, the offerer SHOULD specify which participants are to
respond to a received initiator. A participant that is not specified
SHOULD NOT send responses. Otherwise, undue bandwidth might be
consumed. The offer indicates that each participant that is
specified SHOULD respond if it receives an initiator. It also
indicates that a specified participant MAY send an initiator block.
An SDP answer for a multicast media stream SHOULD include all
collaborative parameters that are present in the offer and that are
supported by the answerer. It SHOULD NOT include any collaborative
parameter that is absent from the offer.
If a participant receives an SDP offer and understands the "rtcp-xr"
attribute but does not wish to implement XR functionality offered,
its answer SHOULD include an "rtcp-xr" attribute without parameters.
By doing so, the party declares that, at a minimum, is capable of
understanding the signaling.
Friedman, et al. Standards Track [Page 41]

RFC 3611 RTCP XR November 20035.3. Usage Outside of Offer/Answer
SDP can be employed outside of the Offer/Answer context, for instance
for multimedia sessions that are announced through the Session
Announcement Protocol (SAP) [15], or streamed through the Real Time
Streaming Protocol (RTSP) [17]. The signaling model is simpler, as
the sender does not negotiate parameters, but the functionality
expected from specifying the "rtcp-xr" attribute is the same as in
Offer/Answer.
When a unilateral parameter is specified for the "rtcp-xr" attribute
associated with a media stream, the receiver of that stream SHOULD
send the corresponding XR block. When a collaborative parameter is
specified, only the participants indicated by the mode value in the
collaborative parameter are concerned. Each such participant that
receives an initiator block SHOULD send the corresponding response
block. Each such participant MAY also send initiator blocks.
6. IANA Considerations
This document defines a new RTCP packet type, the Extended Report
(XR) type, within the existing Internet Assigned Numbers Authority
(IANA) registry of RTP RTCP Control Packet Types. This document also
defines a new IANA registry: the registry of RTCP XR Block Types.
Within this new registry, this document defines an initial set of
seven block types and describes how the remaining types are to be
allocated.
Further, this document defines a new SDP attribute, "rtcp-xr", within
the existing IANA registry of SDP Parameters. It defines a new IANA
registry, the registry of RTCP XR SDP Parameters, and an initial set
of six parameters, and describes how additional parameters are to be
allocated.
6.1. XR Packet Type
The XR packet type defined by this document is registered with the
IANA as packet type 207 in the registry of RTP RTCP Control Packet
types (PT).
6.2. RTCP XR Block Type Registry
This document creates an IANA registry called the RTCP XR Block Type
Registry to cover the name space of the Extended Report block type
(BT) field specified in Section 3. The BT field contains eight bits,
allowing 256 values. The RTCP XR Block Type Registry is to be
managed by the IANA according to the Specification Required policy of
Friedman, et al. Standards Track [Page 42]

RFC 3611 RTCP XR November 2003RFC 2434 [7]. Future specifications SHOULD attribute block type
values in strict numeric order following the values attributed in
this document:
BT name
-- ----
1 Loss RLE Report Block
2 Duplicate RLE Report Block
3 Packet Receipt Times Report Block
4 Receiver Reference Time Report Block
5 DLRR Report Block
6 Statistics Summary Report Block
7 VoIP Metrics Report Block
The BT value 255 is reserved for future extensions.
Furthermore, future specifications SHOULD avoid the value 0. Doing
so facilitates packet validity checking, since an all-zeros field
might commonly be found in an ill-formed packet.
Any registration MUST contain the following information:
- Contact information of the one doing the registration, including
at least name, address, and email.
- The format of the block type being registered, consistent with the
extended report block format described in Section 3.
- A description of what the block type represents and how it shall
be interpreted, detailing this information for each of its fields.
6.3. The "rtcp-xr" SDP Attribute
The SDP attribute "rtcp-xr" defined by this document is registered
with the IANA registry of SDP Parameters as follows:
SDP Attribute ("att-field"):
Attribute name: rtcp-xr
Long form: RTP Control Protocol Extended Report Parameters
Type of name: att-field
Type of attribute: session and media level
Subject to charset: no
Purpose: see Section 5 of this document
Reference: this document
Values: see this document and registrations below
Friedman, et al. Standards Track [Page 43]

RFC 3611 RTCP XR November 2003
The attribute has an extensible parameter field and therefore a
registry for these parameters is required. This document creates an
IANA registry called the RTCP XR SDP Parameters Registry. It
contains the six parameters defined in Section 5.1: "pkt-loss-rle",
"pkt-dup-rle", "pkt-rcpt-times", "stat-summary", "voip-metrics", and
"recv-rtt".
Additional parameters are to be added to this registry in accordance
with the Specification Required policy of RFC 2434 [7]. Any
registration MUST contain the following information:
- Contact information of the one doing the registration, including
at least name, address, and email.
- An Augmented Backus-Naur Form (ABNF) [2] definition of the
parameter, in accordance with the "format-ext" definition of
Section 5.1.
- A description of what the parameter represents and how it shall be
interpreted, both normally and in Offer/Answer.
7. Security Considerations
This document extends the RTCP reporting mechanism. The security
considerations that apply to RTCP reports [9, Appendix B] also apply
to XR reports. This section details the additional security
considerations that apply to the extensions.
The extensions introduce heightened confidentiality concerns.
Standard RTCP reports contain a limited number of summary statistics.
The information contained in XR reports is both more detailed and
more extensive (covering a larger number of parameters). The per-
packet report blocks and the VoIP Metrics Report Block provide
examples.
The per-packet information contained in Loss RLE, Duplicate RLE, and
Packet Receipt Times Report Blocks facilitates multicast inference of
network characteristics (MINC) [11]. Such inference can reveal the
gross topology of a multicast distribution tree, as well as
parameters, such as the loss rates and delays, along paths between
branching points in that tree. Such information might be considered
sensitive to autonomous system administrators.
The VoIP Metrics Report Block provides information on the quality of
ongoing voice calls. Though such information might be carried in an
application specific format in standard RTP sessions, making it
available in a standard format here makes it more available to
potential eavesdroppers.
Friedman, et al. Standards Track [Page 44]

RFC 3611 RTCP XR November 2003
No new mechanisms are introduced in this document to ensure
confidentiality. Encryption procedures, such as those being
suggested for a Secure RTCP (SRTCP) [12] at the time that this
document was written, can be used when confidentiality is a concern
to end hosts. Given that RTCP traffic can be encrypted by the end
hosts, autonomous systems must be prepared for the fact that certain
aspects of their network topology can be revealed.
Any encryption or filtering of XR report blocks entails a loss of
monitoring information to third parties. For example, a network that
establishes a tunnel to encrypt VoIP Report Blocks denies that
information to the service providers traversed by the tunnel. The
service providers cannot then monitor or respond to the quality of
the VoIP calls that they carry, potentially creating problems for the
network's users. As a default, XR packets should not be encrypted or
filtered.
The extensions also make certain denial of service attacks easier.
This is because of the potential to create RTCP packets much larger
than average with the per packet reporting capabilities of the Loss
RLE, Duplicate RLE, and Timestamp Report Blocks. Because of the
automatic bandwidth adjustment mechanisms in RTCP, if some session
participants are sending large RTCP packets, all participants will
see their RTCP reporting intervals lengthened, meaning they will be
able to report less frequently. To limit the effects of large
packets, even in the absence of denial of service attacks,
applications SHOULD place an upper limit on the size of the XR report
blocks they employ. The "thinning" techniques described in Section4.1 permit the packet-by-packet report blocks to adhere to a
predefined size limit.
Friedman, et al. Standards Track [Page 45]

RFC 3611 RTCP XR November 2003A. AlgorithmsA.1. Sequence Number Interpretation
This is the algorithm suggested by Section 4.1 for keeping track of
the sequence numbers from a given sender. It implements the
accounting practice required for the generation of Loss RLE Report
Blocks.
This algorithm keeps track of 16 bit sequence numbers by translating
them into a 32 bit sequence number space. The first packet received
from a source is considered to have arrived roughly in the middle of
that space. Each packet that follows is placed either ahead of or
behind the prior one in this 32 bit space, depending upon which
choice would place it closer (or, in the event of a tie, which choice
would not require a rollover in the 16 bit sequence number).
// The reference sequence number is an extended sequence number
// that serves as the basis for determining whether a new 16 bit
// sequence number comes earlier or later in the 32 bit sequence
// space.
u_int32 _src_ref_seq;
bool _uninitialized_src_ref_seq;
// Place seq into a 32-bit sequence number space based upon a
// heuristic for its most likely location.
u_int32 extend_seq(const u_int16 seq) {
u_int32 extended_seq, seq_a, seq_b, diff_a, diff_b;
if(_uninitialized_src_ref_seq) {
// This is the first sequence number received. Place
// it in the middle of the extended sequence number
// space.
_src_ref_seq = seq | 0x80000000u;
_uninitialized_src_ref_seq = false;
extended_seq = _src_ref_seq;
}
else {
// Prior sequence numbers have been received.
// Propose two candidates for the extended sequence
// number: seq_a is without wraparound, seq_b with
// wraparound.
seq_a = seq | (_src_ref_seq & 0xFFFF0000u);
if(_src_ref_seq < seq_a) {
seq_b = seq_a - 0x00010000u;
diff_a = seq_a - _src_ref_seq;
Friedman, et al. Standards Track [Page 46]

RFC 3611 RTCP XR November 2003
diff_b = _src_ref_seq - seq_b;
}
else {
seq_b = seq_a + 0x00010000u;
diff_a = _src_ref_seq - seq_a;
diff_b = seq_b - _src_ref_seq;
}
// Choose the closer candidate. If they are equally
// close, the choice is somewhat arbitrary: we choose
// the candidate for which no rollover is necessary.
if(diff_a < diff_b) {
extended_seq = seq_a;
}
else {
extended_seq = seq_b;
}
// Set the reference sequence number to be this most
// recently-received sequence number.
_src_ref_seq = extended_seq;
}
// Return our best guess for a 32-bit sequence number that
// corresponds to the 16-bit number we were given.
return extended_seq;
}
A.2. Example Burst Packet Loss Calculation.
This is an algorithm for measuring the burst characteristics for the
VoIP Metrics Report Block (Section 4.7). The algorithm, which has
been verified against a working implementation for correctness, is
reproduced from ETSI TS 101 329-5 [3]. The algorithm, as described
here, takes precedence over any change that might eventually be made
to the algorithm in future ETSI documents.
This algorithm is event driven and hence extremely computationally
efficient.
Given the following definition of states:
state 1 = received a packet during a gap
state 2 = received a packet during a burst
state 3 = lost a packet during a burst
state 4 = lost an isolated packet during a gap
Friedman, et al. Standards Track [Page 47]

RFC 3611 RTCP XR November 2003
The "c" variables below correspond to state transition counts, i.e.,
c14 is the transition from state 1 to state 4. It is possible to
infer one of a pair of state transition counts to an accuracy of 1
which is generally sufficient for this application.
"pkt" is the count of packets received since the last packet was
declared lost or discarded, and "lost" is the number of packets lost
within the current burst. "packet_lost" and "packet_discarded" are
Boolean variables that indicate if the event that resulted in this
function being invoked was a lost or discarded packet.
if(packet_lost) {
loss_count++;
}
if(packet_discarded) {
discard_count++;
}
if(!packet_lost && !packet_discarded) {
pkt++;
}
else {
if(pkt >= gmin) {
if(lost == 1) {
c14++;
}
else {
c13++;
}
lost = 1;
c11 += pkt;
}
else {
lost++;
if(pkt == 0) {
c33++;
}
else {
c23++;
c22 += (pkt - 1);
}
}
pkt = 0;
}
At each reporting interval the burst and gap metrics can be
calculated as follows.
Friedman, et al. Standards Track [Page 48]

RFC 3611 RTCP XR November 2003
// Calculate additional transition counts.
c31 = c13;
c32 = c23;
ctotal = c11 + c14 + c13 + c22 + c23 + c31 + c32 + c33;
// Calculate burst and densities.
p32 = c32 / (c31 + c32 + c33);
if((c22 + c23) < 1) {
p23 = 1;
}
else {
p23 = 1 - c22/(c22 + c23);
}
burst_density = 256 * p23 / (p23 + p32);
gap_density = 256 * c14 / (c11 + c14);
// Calculate burst and gap durations in ms
m = frameDuration_in_ms * framesPerRTPPkt;
gap_length = (c11 + c14 + c13) * m / c13;
burst_length = ctotal * m / c13 - lgap;
/* calculate loss and discard rates */
loss_rate = 256 * loss_count / ctotal;
discard_rate = 256 * discard_count / ctotal;
Intellectual Property Notice
The IETF takes no position regarding the validity or scope of any
intellectual property or other rights that might be claimed to
pertain to the implementation or use of the technology described in
this document or the extent to which any license under such rights
might or might not be available; neither does it represent that it
has made any effort to identify any such rights. Information on the
IETF's procedures with respect to rights in standards-track and
standards-related documentation can be found in BCP 11 [5]. Copies
of claims of rights made available for publication and any assurances
of licenses to be made available, or the result of an attempt made to
obtain a general license or permission for the use of such
proprietary rights by implementors or users of this specification can
be obtained from the IETF Secretariat.
The IETF invites any interested party to bring to its attention any
copyrights, patents or patent applications, or other proprietary
rights which may cover technology that may be required to practice
this standard. Please address the information to the IETF Executive
Director.
Friedman, et al. Standards Track [Page 49]

RFC 3611 RTCP XR November 2003
Full Copyright Statement
Copyright (C) The Internet Society (2003). All Rights Reserved.
This document and translations of it may be copied and furnished to
others, and derivative works that comment on or otherwise explain it
or assist in its implementation may be prepared, copied, published
and distributed, in whole or in part, without restriction of any
kind, provided that the above copyright notice and this paragraph are
included on all such copies and derivative works. However, this
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Internet organizations, except as needed for the purpose of
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The limited permissions granted above are perpetual and will not be
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This document and the information contained herein is provided on an
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HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
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Acknowledgement
Funding for the RFC Editor function is currently provided by the
Internet Society.
Friedman, et al. Standards Track [Page 55]